The present invention relates to a method of controlling a hearing instrument based on identifying an acoustic environment, and a corresponding hearing instrument.
It is common for state-of-the-art hearing instruments to incorporate automatic control of actuators such as noise cancellers, beam formers, and so on, to automatically adjust the hearing instrument to optimise the sound output for the wearer dependent on the acoustic environment. Such automatic control is based on classifying types of acoustic environments into broad classes, such as “clean speech”, “speech in noise”, “noise”, and “music”. This is typically achieved by processing sound information from a microphone and extracting characteristic features of the sound information, such as energy spectrums, frequency responses, signal-to-noise ratios, signal directions, and so on. Based on the result of the extraction of characteristic features, parameters of the audio signal processing unit are adjusted to optimise the wearer's hearing experience in his or her present surroundings. This optimisation can be by means of predefined programs, or adjusting individual parameters as required.
With various prior art systems, there are several limitations: the detectable classes of acoustic environments are rather broad, leading to insufficient hearing performance for some specific hearing scenarios; extra hardware is often required, increasing costs, power consumption and complexity; and many of the prior art solutions rely on real-time communication between hearing devices and sometimes also a beacon or other separate module that has to be carried by the wearer. Real-time communication uses a lot of power, leading to short battery life and frequent battery changes.